This change rewrites RMSLevel, making it accept an ArrayView as input,
and modify the implementation somewhat. It also makes the class keep
track of the peak RMS in addition to the average RMS over the
measurement period.
New tests are added to cover the new functionality.
BUG=webrtc:6622
Review-Url: https://codereview.webrtc.org/2535523002
Cr-Commit-Position: refs/heads/master@{#15294}
Remove entries for headers that no longer exists in the webrtc/ dir.
BUG=webrtc:5878
NOTRY=True
Review-Url: https://codereview.webrtc.org/2540633002
Cr-Commit-Position: refs/heads/master@{#15292}
Issue: This API was calculating the file_header and frame_header offset only for the first frame which is not the right logic. We need to skip the file and frame header every time we extract a new frame.
Also added a unit test case which compares the extracted frame with the frame stored in text file.
NOPRESUBMIT=true
NOTRY=true
BUG=webrtc:6761
Review-Url: https://codereview.webrtc.org/2532963002
Cr-Commit-Position: refs/heads/master@{#15288}
This is an integration test using peerconnectiontestwrapper.h to set up
and end to end test using a real PeerConnection implementation. These
tests will complement rtcstatscollector_unittest.cc which collects all
stats using mocks.
The integration test is set up so that all stats types are returned by
GetStats and verifies that expected dictionary members are defined. The
test could in the future be updated to include sanity checks for the
values of members. There is a sanity check that references to other
stats dictionaries yield existing stats of the appropriate type, but
other than that members are only tested for if they are defined not.
StatsCallback of rtcstatscollector_unittest.cc is moved so that it can
be reused and renamed to RTCStatsObtainer.
TODO: Audio stream track stats members are missing in the test. Find out
if this is because of a real problem or because of testing without real
devices. Do this before closing crbug.com/627816.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2521663002
Cr-Commit-Position: refs/heads/master@{#15287}
This brings QualityScaler much more in line with OveruseFrameDetector.
The two classes are conceptually similar, and should be used in the
same way. The biggest changes in this CL are:
- Quality scaling is now only done in ViEEncoder and not in each
encoder implementation separately.
- QualityScaler now checks the average QP asynchronously, instead of
having to be polled on each frame.
- QualityScaler is no longer responsible for actually scaling the frames,
but has a callback to ViEEncoder that it uses to express it's desire
for lower resolution.
BUG=webrtc:6495
Review-Url: https://codereview.webrtc.org/2398963003
Cr-Commit-Position: refs/heads/master@{#15286}
Intervals when video is paused is no longer included in the stats:
"WebRTC.Video.InputFramesPerSecond"
"WebRTC.Video.SentFramesPerSecond"
BUG=webrtc:5283
Review-Url: https://codereview.webrtc.org/2536743002
Cr-Commit-Position: refs/heads/master@{#15285}
The actual time used in production code should honor the epoch time.
BUG=webrtc:6737
Review-Url: https://codereview.webrtc.org/2526433002
Cr-Commit-Position: refs/heads/master@{#15282}
There's no longer any need to make the two arguments have the same
signedness, so we can remove a bunch of superfluous (and sometimes
dangerous) casts.
It turned out I also had to fix the safe_cmp functions to properly handle
enums that are implicitly convertible to integers.
NOPRESUBMIT=true
BUG=webrtc:6645
Review-Url: https://codereview.webrtc.org/2534683002
Cr-Commit-Position: refs/heads/master@{#15281}
There's no longer any need to make the two arguments have the same
signedness, so we can drop the "u" suffix on literal integer
arguments.
NOPRESUBMIT=true
BUG=webrtc:6645
Review-Url: https://codereview.webrtc.org/2535593002
Cr-Commit-Position: refs/heads/master@{#15280}
Now ProbeController can send periodic bandwidth probes when in
application-limited region. This will allow to maintain correct
bottleneck bandwidth estimate, even not all bandwidth is being used.
The feature is not enabled by default, but can be enabled with a flag.
Interval between probes is currently set to 5 seconds.
BUG=webrtc:6332
Review-Url: https://codereview.webrtc.org/2504023002
Cr-Commit-Position: refs/heads/master@{#15279}
Previously, a frame queued before calling addFrameListener could be
passed to the listener. Also fixes an issue where listener could still
be called after removeFrameListener call returned.
BUG=webrtc:6470
Review-Url: https://codereview.webrtc.org/2529313002
Cr-Commit-Position: refs/heads/master@{#15275}
In this CL:
- EndToEndTests is now parameterized.
- Added VP8 non-rotated unittest.
- CanReceiveUlpfec/CanReceiveFlexFec now use multisets for timestamps.
- pre_decode_image_callback_ is now called before decoding a frame
with the new video jitter buffer.
- Set video rotation when FrameObjects are created.
- Calculate KeyFramesReceivedInPermille in new video jitter buffer.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2522493002
Cr-Commit-Position: refs/heads/master@{#15274}
This logic doesn't really work. Application should mask the view while
the surface size is being changed.
BUG=webrtc:6470
Review-Url: https://codereview.webrtc.org/2528243003
Cr-Commit-Position: refs/heads/master@{#15273}
Reason for revert:
Unfortunately, this change breaks internal projects. Specifically the change to the CongestionController interface means anything implementing it will be forced to change in lock-step.
Original issue's description:
> Pass time constanct to bwe smoothing filter.
>
> BUG=webrtc:6443, webrtc:6303
>
> Committed: https://crrev.com/9abbf5ae4ec7d688a9b4aa03a405f3faadb74b90
> Cr-Commit-Position: refs/heads/master@{#15266}
TBR=minyue@webrtc.org,stefan@webrtc.org,solenberg@webrtc.org,michaelt@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6443, webrtc:6303
Review-Url: https://codereview.webrtc.org/2532993002
Cr-Commit-Position: refs/heads/master@{#15272}
Pass the selected cricket::VideoCodec to H264EncoderImpl::H264EncoderImpl. The cricket::VideoCodec contains relevant information for H264 about selected profile and packetization mode.
BUG=chromium:600254,webrtc:6402, webrtc:6337
Review-Url: https://codereview.webrtc.org/2474993002
Cr-Commit-Position: refs/heads/master@{#15270}
Not needed since it doesn't run in a device. It will enable us to run JUnit
tests on android on swarming too :) and more good stuff like flakiness dashboard
BUG=chromium:497757
R=kjellander@webrtc.org
Review-Url: https://codereview.webrtc.org/2531993003
Cr-Commit-Position: refs/heads/master@{#15268}
transport.h defines an interface for sending rtp and rtcp packets,
which is used by MediaChannel in webrtc/media/engine,
{Audio|Video}{Send|Receive}Stream and in a few other
places. It was part of the build target //webrtc:webrtc, which is a monolithic target with
all webrtc production code. This CL moves the header to its own target in webrtc/api
and deprecates the old location.
Targets in webrtc/api should in general only depend on other
targets in webrtc/api. The target webrtc/api:call_api depends on
transport.h. This change also makes webrtc/voice_engine pass GN's header
include checker and is needed in order for webrtc/api:call_api to pass
it.
transport.h will be completely removed in a follow-up CL in a few weeks
after clients have updated their includes.
NOTRY=True
BUG=webrtc:5589, webrtc:5878, webrtc:6785
Review-Url: https://codereview.webrtc.org/2426563003
Cr-Commit-Position: refs/heads/master@{#15267}
A WebRtcVideoEngine2 object seems to be reused between PeerConnections,
which means that the field trial added in
https://codereview.webrtc.org/2511703002/ may not activate/deactivate
as intended between calls. This CL removes the caching of video codecs,
which gets rid of this problem.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2521393004
Cr-Commit-Position: refs/heads/master@{#15265}
Reason for revert:
Breaking some trybots due to memory error.
Original issue's description:
> Fixed bug in ExtractFrameFromY4mFile API which was not extracting the frames correctly.
>
> Issue: This API was calculating the file_header and frame_header offset only for the first frame which is not the right logic. We need to skip the file and frame header every time we extract a new frame.
>
> Also added a unit test case which compares the extracted frame with the frame stored in text file.
>
> BUG=webrtc:6761
>
> NOPRESUBMIT=true
> NOTRY=true
>
> Committed: https://crrev.com/b7636b4656d7f8c368963f2256dc2ef7b7ba89c8
> Cr-Commit-Position: refs/heads/master@{#15260}
TBR=phoglund@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6761
Review-Url: https://codereview.webrtc.org/2535783002
Cr-Commit-Position: refs/heads/master@{#15262}
Issue: This API was calculating the file_header and frame_header offset only for the first frame which is not the right logic. We need to skip the file and frame header every time we extract a new frame.
Also added a unit test case which compares the extracted frame with the frame stored in text file.
BUG=webrtc:6761
NOPRESUBMIT=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2529923002
Cr-Commit-Position: refs/heads/master@{#15260}
This function has no public use,
removed tests calling it: effect of registering extension is better
tested in AllocatePacket and SendPacket tests.
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2530363002
Cr-Commit-Position: refs/heads/master@{#15258}
Reason for revert:
Include fix; set profile information in CreatePayloadType for video.
Original issue's description:
> Revert of Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload (patchset #1 id:40001 of https://codereview.webrtc.org/2525693003/ )
>
> Reason for revert:
> The CL doesn't actually set profile information in VideoPayload.
>
> Original issue's description:
> > Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload
> >
> > It's necessary to check H264 profile information as well as payload name
> > in PayloadIsCompatible in RTPPayloadRegistry.
> >
> > BUG=webrtc:6743
> >
> > Committed: https://crrev.com/bdbc4b7ef578ba1d61ceec351bc47c33da115329
> > Cr-Commit-Position: refs/heads/master@{#15248}
>
> TBR=mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6743
>
> Committed: https://crrev.com/d7e6ccbc53fc24acdcc7507a6f3a155626473d54
> Cr-Commit-Position: refs/heads/master@{#15251}
TBR=mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6743
Review-Url: https://codereview.webrtc.org/2529153002
Cr-Commit-Position: refs/heads/master@{#15252}
Reason for revert:
The CL doesn't actually set profile information in VideoPayload.
Original issue's description:
> Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload
>
> It's necessary to check H264 profile information as well as payload name
> in PayloadIsCompatible in RTPPayloadRegistry.
>
> BUG=webrtc:6743
>
> Committed: https://crrev.com/bdbc4b7ef578ba1d61ceec351bc47c33da115329
> Cr-Commit-Position: refs/heads/master@{#15248}
TBR=mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6743
Review-Url: https://codereview.webrtc.org/2529143002
Cr-Commit-Position: refs/heads/master@{#15251}
A recent cl (https://codereview.webrtc.org/2510583002) introduced an
issue where temporal layers may return incorrect bitrates, given that
they are stateful and that the GetPreferredBitrateBps is called.
The fix is to use a temporary simulcast rate allocator instance, without
temporal layers, and get the preferred bitrate from that.
Additionally, some regression in bitrate allocated stems from overly
often reconfiguring the encoder, which yields suboptimal rate control.
The fix here is to limit encoder updates to when values have actually
changed.
As a bonus, dchecks added by this cl found a bug in the (unused) RealtimeTemporalLayers implementation. Fixed that as well.
BUG=webrtc:6301, chromium:666654
Review-Url: https://codereview.webrtc.org/2529073003
Cr-Commit-Position: refs/heads/master@{#15250}
It's necessary to check H264 profile information as well as payload name
in PayloadIsCompatible in RTPPayloadRegistry.
BUG=webrtc:6743
Review-Url: https://codereview.webrtc.org/2525693003
Cr-Commit-Position: refs/heads/master@{#15248}
The Android HW encoder is currently not setting any H264 codec parameters or profile information. No profile-level-id means Baseline Level 1, but we are actually using Contrained Baseline Level 3.1. This CL sets the correct codec parameters.
BUG=webrtc:6337
Review-Url: https://codereview.webrtc.org/2497163002
Cr-Commit-Position: refs/heads/master@{#15247}
